I. Field of the Invention
The present invention pertains generally to the field of speech processing, and more specifically to methods and apparatus for quantizing line spectral information in speech coders.
II. Background
Transmission of voice by digital techniques has become widespread, particularly in long distance and digital radio telephone applications. This, in turn, has created interest in determining the least amount of information that can be sent over a channel while maintaining the perceived quality of the reconstructed speech. If speech is transmitted by simply sampling and digitizing, a data rate on the order of sixty-four kilobits per second (kbps) is required to achieve a speech quality of conventional analog telephone. However, through the use of speech analysis, followed by the appropriate coding, transmission, and resynthesis at the receiver, a significant reduction in the data rate can be achieved.
Devices for compressing speech find use in many fields of telecommunications. An exemplary field is wireless communications. The field of wireless communications has many applications including, e.g., cordless telephones, paging, wireless local loops, wireless telephony such as cellular and PCS telephone systems, mobile Internet Protocol (IP) telephony, and satellite communication systems. A particularly important application is wireless telephony for mobile subscribers.
Various over-the-air interfaces have been developed for wireless communication systems including, e.g., frequency division multiple access (FDMA), time division multiple access (TDMA), and code division multiple access (CDMA). In connection therewith, various domestic and international standards have been established including, e.g., Advanced Mobile Phone Service (AMPS), Global System for Mobile Communications (GSM), and Interim Standard 95 (IS-95). An exemplary wireless telephony communication system is a code division multiple access (CDMA) system. The IS-95 standard and its derivatives, IS-95A, ANSI J-STD-008, IS-95B, proposed third generation standards IS-95C and IS2000, etc. (referred to collectively herein as IS-95), are promulgated by the Telecommunication Industry Association (TIA) and other well known standards bodies to specify the use of a CDMA over-the-air interface for cellular or PCS telephony communication systems. Exemplary wireless communication systems configured substantially in accordance with the use of the IS-95 standard are described in U.S. Pat. Nos. 5,103,459 and 4,901,307, which are assigned to the assignee of the present invention and fully incorporated herein by reference.
Devices that employ techniques to compress speech by extracting parameters that relate to a model of human speech generation are called speech coders. A speech coder divides the incoming speech signal into blocks of time, or analysis frames. Speech coders typically comprise an encoder and a decoder. The encoder analyzes the incoming speech frame to extract certain relevant parameters, and then quantizes the parameters into binary representation, i.e., to a set of bits or a binary data packet. The data packets are transmitted over the communication channel to a receiver and a decoder. The decoder processes the data packets, unquantizes them to produce the parameters, and resynthesizes the speech frames using the unquantized parameters.
The function of the speech coder is to compress the digitized speech signal into a low-bit-rate signal by removing all of the natural redundancies inherent in speech. The digital compression is achieved by representing the input speech frame with a set of parameters and employing quantization to represent the parameters with a set of bits. If the input speech frame has a number of bits Ni and the data packet produced by the speech coder has a number of bits No, the compression factor achieved by the speech coder is Cr=Ni/No. The challenge is to retain high voice quality of the decoded speech while achieving the target compression factor. The performance of a speech coder depends on (1) how well the speech model, or the combination of the analysis and synthesis process described above, performs, and (2) how well the parameter quantization process is performed at the target bit rate of No bits per frame. The goal of the speech model is thus to capture the essence of the speech signal, or the target voice quality, with a small set of parameters for each frame.
Perhaps most important in the design of a speech coder is the search for a good set of parameters (including vectors) to describe the speech signal. A good set of parameters requires a low system bandwidth for the reconstruction of a perceptually accurate speech signal. Pitch, signal power, spectral envelope (or formants), amplitude and phase spectra are examples of the speech coding parameters.
Speech coders may be implemented as time-domain coders, which attempt to capture the time-domain speech waveform by employing high time resolution processing to encode small segments of speech (typically 5 millisecond (ms) subframes) at a time. For each subframe, a high-precision representative from a codebook space is found by means of various search algorithms known in the art. Alternatively, speech coders may be implemented as frequency-domain coders, which attempt to capture the short-term speech spectrum of the input speech frame with a set of parameters (analysis) and employ a corresponding synthesis process to recreate the speech waveform from the spectral parameters. The parameter quantizer preserves the parameters by representing them with stored representations of code vectors in accordance with known quantization techniques described in A. Gersho and R. M. Gray, Vector Quantization and Signal Compression (1992).
A well-known time-domain speech coder is the Code Excited Linear Predictive (CELP) coder described in L. B. Rabiner and R. W. Schafer, Digital Processing of Speech Signals 396-453 (1978), which is fully incorporated herein by reference. In a CELP coder, the short term correlations, or redundancies, in the speech signal are removed by a linear prediction (LP) analysis, which finds the coefficients of a short-term formant filter. Applying the short-term prediction filter to the incoming speech frame generates an LP residue signal, which is further modeled and quantized with long-term prediction filter parameters and a subsequent stochastic codebook. Thus, CELP coding divides the task of encoding the time-domain speech waveform into the separate tasks of encoding the LP short-term filter coefficients and encoding the LP residue. Time-domain coding can be performed at a fixed rate (i.e., using the same number of bits, No, for each frame) or at a variable rate (in which different bit rates are used for different types of frame contents). Variable-rate coders attempt to use only the amount of bits needed to encode the code parameters to a level adequate to obtain a target quality. An exemplary variable rate CELP coder is described in U.S. Pat. No. 5,414,796, which is assigned to the assignee of the present invention and fully incorporated herein by reference.
Time-domain coders such as the CELP coder typically rely upon a high number of bits, No, per frame to preserve the accuracy of the time-domain speech waveform. Such coders typically deliver excellent voice quality provided the number of bits, No, per frame relatively large (e.g., 8 kbps or above). However, at low bit rates (4 kbps and below), time-domain coders fail to retain high quality and robust performance due to the limited number of available bits. At low bit rates, the limited codebook space clips the waveform matching capability of conventional time-domain coders, which are so successfully deployed in higher-rate commercial applications. Hence, despite improvements over time, many CELP coding systems operating at low bit rates suffer from perceptually significant distortion typically characterized as noise.
There is presently a surge of research interest and strong commercial need to develop a high-quality speech coder operating at medium to low bit rates (i.e., in the range of 2.4 to 4 kbps and below). The application areas include wireless telephony, satellite communications, Internet telephony, various multimedia and voice-streaming applications, voice mail, and other voice storage systems. The driving forces are the need for high capacity and the demand for robust performance under packet loss situations. Various recent speech coding standardization efforts are another direct driving force propelling research and development of low-rate speech coding algorithms. A low-rate speech coder creates more channels, or users, per allowable application bandwidth, and a low-rate speech coder coupled with an additional layer of suitable channel coding can fit the overall bit-budget of coder specifications and deliver a robust performance under channel error conditions.
One effective technique to encode speech efficiently at low bit rates is multimode coding. An exemplary multimode coding technique is described in U.S. application Ser. No. 09/217,341, entitled VARIABLE RATE SPEECH CODING, filed Dec. 21, 1998, assigned to the assignee of the present invention, and fully incorporated herein by reference. Conventional multimode coders apply different modes, or encoding-decoding algorithms, to different types of input speech frames. Each mode, or encoding-decoding process, is customized to optimally represent a certain type of speech segment, such as, e.g., voiced speech, unvoiced speech, transition speech (e.g., between voiced and unvoiced), and background noise (nonspeech) in the most efficient manner. An external, open-loop mode decision mechanism examines the input speech frame and makes a decision regarding which mode to apply to the frame. The open-loop mode decision is typically performed by extracting a number of parameters from the input frame, evaluating the parameters as to certain temporal and spectral characteristics, and basing a mode decision upon the evaluation.
In many conventional speech coders, line spectral information such as line spectral pairs or line spectral cosines is transmitted without exploiting the steady-state nature of voiced speech by encoding voiced speech frames without reducing the coding rate sufficiently. Hence, valuable bandwidth is wasted. In other conventional speech coders, multimode speech coders, or low-bit-rate speech coders, the steady-state nature of voiced speech is exploited for every frame. Accordingly, nonsteady-state frames degrade, and voice quality suffers.
The present invention is directed to a speech coder that uses multiple vector quantization methods to adapt to changes between periodic frames and nonperiodic frames. Accordingly, in one aspect of the invention, a speech coder advantageously includes a linear predictive filter configured to analyze a frame and generate a line spectral information codevector based thereon; and a quantizer coupled to the linear predictive filter and configured to vector quantize the line spectral information vector with a first vector quantization technique that uses a non-moving-average prediction-based vector quantization scheme, wherein the quantizer is further configured to compute equivalent moving average codevectors for the first technique, update with the equivalent moving average codevectors a memory of a moving average codebook of codevectors for a predefined number of frames that were previously processed by the speech coder, compute a target quantization vector for the second technique based on the updated moving average codebook memory, vector quantize the target quantization vector with a second vector quantization technique to generate a quantized target codevector, the second vector quantization technique using a moving-average predictionbased scheme, update the memory of the moving average codebook with the quantized target codevector, and compute quantized line spectral information vectors from the quantized target codevector.
It would be advantageous to provide an adaptive coding method that reacts to the nature of the speech content of each frame. Additionally, as the speech signal is generally nonsteady-state, or nonstationary, the efficiency of quantization of the line spectral information (LSI) parameters used in speech coding could be improved by employing a scheme in which the LSI parameters of each frame of speech are selectively coded either using moving-average (MA) prediction-based vector quantization (VQ) or using other standard VQ methods. Such a scheme would suitably exploit the advantages of either of the above two methods of VQ. Hence, it would be desirable to provide a speech coder that interleaves the two methods of VQ by appropriately mixing the two schemes at the boundaries of transitions from one method to the other. Thus, there is a need for a speech coder that uses multiple vector quantization methods to adapt to changes between periodic frames and nonperiodic frames.
The present invention is directed to a speech coder that uses multiple vector quantization methods to adapt to changes between periodic frames and nonperiodic frames. Accordingly, in one aspect of the invention, a speech coder advantageously includes a linear predictive filter configured to analyze a frame and generate a line spectral information codevector based thereon; and a quantizer coupled to the linear predictive filter and configured to vector quantize the line spectral information vector with a first vector quantization technique that uses a non-moving-average prediction-based vector quantization scheme, wherein the quantizer is further configured to compute equivalent moving average codevectors for the first technique, update with the equivalent moving average codevectors a memory of a moving average codebook of codevectors for a predefined number of frames that were previously processed by the speech coder, compute a target quantization vector for the second technique based on the updated moving average codebook memory, vector quantize the target quantization vector with a second vector quantization technique to generate a quantized target codevector, the second vector quantization technique using a moving-average prediction-based scheme, update the memory of the moving average codebook with the quantized target codevector, and compute quantized line spectral information vectors from the quantized target codevector.
In another aspect of the invention, a method of vector quantizing a line spectral information vector of a frame, using first and second quantization vector quantization techniques, the first technique using a non-moving-average prediction-based vector quantization scheme, the second technique using a moving-average prediction-based vector quantization scheme, advantageously includes the steps of vector quantizing the line spectral information vector with the first vector quantization technique; computing equivalent moving average codevectors for the first technique; updating with the equivalent moving average codevectors a memory of a moving average codebook of codevectors for a predefined number of frames that were previously processed by the speech coder; calculating a target quantization vector for the second technique based on the updated moving average codebook memory; vector quantizing the target quantization vector with the second vector quantization technique to generate a quantized target codevector; updating the memory of the moving average codebook with the quantized target codevector; and deriving quantized line spectral information vectors from the quantized target codevector.
In another aspect of the invention, a speech coder advantageously includes means for vector quantizing a line spectral information vector of a frame with a first vector quantization technique that uses a non-movingaverage prediction-based vector quantization scheme; means for computing equivalent moving average codevectors for the first technique; means for updating with the equivalent moving average codevectors a memory of a moving average codebook of codevectors for a predefined number of frames that were previously processed by the speech coder; means for calculating a target quantization vector for the second technique based on the updated moving average codebook memory; means for vector quantizing the target quantization vector with the second vector quantization technique to generate a quantized target codevector; means for updating the memory of the moving average codebook with the quantized target codevector; and means for deriving quantized line spectral information vectors from the quantized target codevector.